From f3e2d68df62d3ef13759d76bb1d87903d08ee586 Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Sat, 25 Aug 2012 13:04:33 -0400 Subject: aacenc: use planar sample format --- libavcodec/aacenc.c | 33 +++++++++++++++------------------ 1 file changed, 15 insertions(+), 18 deletions(-) (limited to 'libavcodec/aacenc.c') diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c index f7fbde2694..f5ab40f459 100644 --- a/libavcodec/aacenc.c +++ b/libavcodec/aacenc.c @@ -479,31 +479,28 @@ static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, } /* - * Deinterleave input samples. + * Copy input samples. * Channels are reordered from Libav's default order to AAC order. */ -static void deinterleave_input_samples(AACEncContext *s, const AVFrame *frame) +static void copy_input_samples(AACEncContext *s, const AVFrame *frame) { - int ch, i; - const int sinc = s->channels; - const uint8_t *channel_map = aac_chan_maps[sinc - 1]; + int ch; + int end = 2048 + (frame ? frame->nb_samples : 0); + const uint8_t *channel_map = aac_chan_maps[s->channels - 1]; - /* deinterleave and remap input samples */ - for (ch = 0; ch < sinc; ch++) { + /* copy and remap input samples */ + for (ch = 0; ch < s->channels; ch++) { /* copy last 1024 samples of previous frame to the start of the current frame */ memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0])); - /* deinterleave */ - i = 2048; + /* copy new samples and zero any remaining samples */ if (frame) { - const float *sptr = ((const float *)frame->data[0]) + channel_map[ch]; - for (; i < 2048 + frame->nb_samples; i++) { - s->planar_samples[ch][i] = *sptr; - sptr += sinc; - } + memcpy(&s->planar_samples[ch][2048], + frame->extended_data[channel_map[ch]], + frame->nb_samples * sizeof(s->planar_samples[0][0])); } - memset(&s->planar_samples[ch][i], 0, - (3072 - i) * sizeof(s->planar_samples[0][0])); + memset(&s->planar_samples[ch][end], 0, + (3072 - end) * sizeof(s->planar_samples[0][0])); } } @@ -526,7 +523,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, return ret; } - deinterleave_input_samples(s, frame); + copy_input_samples(s, frame); if (s->psypp) ff_psy_preprocess(s->psypp, s->planar_samples, s->channels); @@ -826,7 +823,7 @@ AVCodec ff_aac_encoder = { .close = aac_encode_end, .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLT, + .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), .priv_class = &aacenc_class, -- cgit v1.2.3